With the rapid development of manufacturing techniques for the super-large-scale integrated circuit, the trend for integrating and digitizing of electro-acoustical products has become increasingly evident, and the design and manufacture of the speaker system—as leading products of the electro-acoustic industry, gradually develop towards a direction of a low power consumption, miniaturization and portable type. Reviewing the development process of speaker systems, it can be divided into three stages: the analog speaker system, the semi-digital speaker system and the digital speaker system. The traditional analog speaker system has been no longer welcomed by the broad consumers due to the problems such as the low electro-acoustic conversion efficiency, high power consumption and heat, and the large volume and weight and the like; on the contrary, in recent years, the semi-digital speaker system generated by the digitization wave driving, because of to the use of pulse width modulation (PWM) or Δ-Σ modulation (Delta-Sigma modulation) and Class D power amplifying drive technology, successfully resolves the problem of power consumption and heat, greatly enhances the electro-acoustic conversion efficiency of the entire system, and accordingly can achieve a miniaturization level, which prompts the wide use of the semi-digital speaker system in the multimedia sound boxes, mobile phones, mp3 players, digital cameras and laptop computers and other fields. However, the backward stage of the semi-level digital speaker system still needs to rely on a bulky LC low-pass analog filter to filter the out-of-band high frequency component of the digital pulse modulated signal off, to demodulate the modulated low-frequency envelope signal, and thereby to complete the digital-analog conversion process. These semi-digital speaker systems have promoted the digitalization of the system to a power-amplifying stage, however, between the power amplifier and the speaker unit, it usually needs to rely on an analog low-pass filter consisting of inducers and capacitors to complete a digital-to-analog conversion, so as to ensure the speaker unit is in an analog input state; in addition, on the current market, many chip companies has launched digital power amplifier chips without analog low-pass LC filters, but these power amplifier chips do not consider an uniformly digital encoding processing by taking multiple speaker units or multiple voice coils as an entirety, and have a low performance and a limited suppression ability in terms of noise and harmonic distortion suppression, and meanwhile these power amplifier chips are restricted to drive a small-caliber speaker unit of a few watts order; for digitization driving of high-power speaker units, it still can not get rid of restrictions of the analog LC filter.
In order to eliminate the restrictions of the analog LC filter, to breakthrough the digitization bottleneck of speaker units, to improve the integration level of the speaker systems, to achieve a complete digitization of all signal processing and transmission steps in a speaker system, it is required to include the speaker unit into the digital coding step, to really achieve digitized coding of the speaker unit, to form a digitized speaker system, and thereby ultimately due to the low-pass filtering characteristic of the speaker unit and the human ear structure, converting from a digital coding vector to an analog vibration vector is completed, the digital-to-analog conversion step is transferred to be achieved in a physical stage of electro-acoustic transducing, and thereby a digital-to-analog conversion device included by a conventional system is taken away, and a variety of electrical noises introduced by digital-to-analog converter are avoided.
The digital to analog conversion process of the digital speaker system no longer relys on the traditional digital-analog converter chip to achieve, instead by means of the actual physical role of the speaker unit itself in the electro-acoustic conversion process to complete the digital to analog conversion. The speaker loads used by the digital systems usually are two kinds: Digital Speaker Array (DLA) and Multiple Voice Coil Digital Loudspeaker (MVCDL). For the case of DLA load, the digital-analog conversion process is as follow: firstly, each speaker unit independently completes the electro-acoustic conversion—converting a switch electrical signal sent by digital coding into an analog sound signal and independently radiating it into the air, and the electro-acoustic conversion process of each speaker unit is similar to the low-pass filtering, the speaker units also filtering process quantizing noise during the independent filtering process of the digitally coded signal; and then analog sound fields independently radiated by each unit complete a coupled superposition in the air, and thus the analog signal source component is accurately synthesized on the basis of ensuring the cancellation of the analog quantizing noise components radiated by each units. For the case of MVCDL load, the digital-analog conversion process is as follow: firstly, in the process of each winding receiving a digital current signal from the digitally coder, power conversion can occur independently, to convert the digital current signal into an electromagnetic driving signal in the pulse form for driving each windings; and multiple voice coils within the constraints of the binding action itself will coupling superimpose the electromagnetic driving component in the pulse form forced on itself, to form an electromagnetic driving resultant force in an analog form for driving a plurality of windings and the cone to move, thereby promoting the air to vibrate and reproduce an analog sound field. In both load cases, the reproduced analog sound field can be further improved through the low-pass filtering effect of human ear. Centred on the core issue of digitalization of the speaker unit, in recent years, scholars of a number of domestic and overseas research institutions have undertaken relatively wide and deep theoretical and practical researches on digitizing coded modulation technology, digitizing power drive technology and digitizing speaker unit manufacturing technology, and therefore forming a new research field of which the research direction is digitized speaker system design.
Since 1920s, experts and scholars of a number of foreign research institutions carried out theoretical and experimental researches on digital speaker system in succession, and have gained a fruitful research results. These studies mainly embodied on an innovative research of three core technologies, digitalized coding modulation technology, digitalized power amplifying technology and digitalized speaker manufacturing technology. According to the different digital coding systems, the digital speaker systems can be divided into three categories: PCM (Pulse Code Mouldation) coding-based digital speaker systems, 1-bit PWM (Pulse Width Mouldation) coding-based or Δ-Σ (Delta-Sigma Mouldation-DSM) coding-based digital speaker systems, and multi-bit Δ-Σ coding-based digital speaker system.
In 1963, C. Roberts filed the world's first invention patent (U.S. Pat. No. 3,153,229) of a PCM coding-based digital exciter. In 1979, Flanagan put forward a digitalization design on an electret speaker as well as an associated design method of an acoustic low-pass filter facing the application requirement of phones and headsets (J. L. Flanagan. Direct digital conversion in acoustic transducers [J]. J. Acoust. Soc. Am. Suppl. 1, 1979, 66: S54.). In 1977, the Japanese company SONY produced the world's first digital multiple voice coil speaker by a design method of controlling the number of turns of each windings units to increase according to a multiple of an exponential of 2, and developed an associated driving device (Patent No. JP 52121316). In 1986, Nieuwendi jk et from U.S. company PHILIPS improved the winding method of voice coils of the multiple voice coil speaker presented by SONY Corporation in early, and proposed that under the condition of maintaining the same number of turns of each voice coil, to manufacture each voice coil units by successively increasing the numbers of the winded wires of each voice coil according to a multiple of an exponential of 2, and winding a plurality of wires in parallel (U.S. Pat. No. 4,612,421).
These PCM coding-based digital loudspeaker system, mainly focusing on changing the load structure of the speaker to meet the digitalization requirements—designing the radiation area of each vibration unit or the winding turn number of each windings to keep a relationship of a multiple of an exponential of 2, the critical defect existing in the design idea is that simply relying on the increase of vibration unit area or the number of turns of coils to achieve digital system, will cause an increased weight of vibration components, a decreased speaker sensitivity, an enlarged power amplifier driving power, and a reduced electro-acoustic conversion efficiency; increased difficulty and cost in fabrication of components, a decreased speaker yield; a large volume and high weight of the speaker and the power amplifier which is difficult to meet the demand of portability. With the rapid development of electronic technology, started from 1997, Kishigami et al. from Japanese company SONY (U.S. Pat. No. 5,862,237) and Ken ji, et al. from Shinshu University (A. Hayama and the K. Furihata. Acoustic characteristics of an electrodynamic planar digital loudspeaker using noise shaping technology [J]. J. Acoust. Soc. Am., 2005, 117 (6): 3636-3644) began to focus on another way to realize the digital system—controlling the drive current of the power amplifying circuit of each vibration unit (the plate electrode, the piezoelectric patch or the voice coil) to increase according to a relationship of a multiple of an exponential of 2, considering the design of digital loudspeakers from the angle of the digitalization of the power amplifier driving circuit, and thus making up for the design defect of the speaker unit digitalization.
The PCM coding-based digital system requires to correspondingly structurally design the speaker diaphragm area or the number of turns of the voice coil or control the magnitude of the power amplifier driving current in combination with the place value of code, so as to ensure the synthesized analog signals of multiple bits has a good reproduction quality, it significantly increases the design complexity of the speaker or power amplifier, due to that it is hard to precisely control the scaling relation and the values of the diaphragm area, the number of turns of the voice coils and the power amplifier current, accordingly resulting in that it is hard to obtain a well reproduction effect for the PCM coding-based digital system.
The difficult problem existing in the PCM coding-based digital loudspeaker system is that it is difficult to accurately control the manufacture of the digital speaker unit and the drive current strength, and due to the constraints of the above mentioned problem, the PCM coding-based digital system has not been able to obtain a satisfactory level of sound quality. In order to overcome the defects and shortcomings of the PCM coding in the manufacture and the drive control of the digital loudspeaker system, in recent years, many scholars began to study the digital loudspeaker systems employing the 1-bit PWM or Δ-Σ modulation technology, and achieve a series of important research results.
Since 1994, the development team from UK company 1 LIMITED led by their president Doctor Tony Hooley, applied for a series of patents related to 1-bit PWM coded digital loudspeaker system (Patents WO 01/23104 A2 and GB 2373956 A). The realization method of this PWM technology based digital loudspeaker system has two shortcomings: D coding method based on the PWM technology, due to the modulation structure itself has an inherent nonlinear defect, will cause the coded signal to generate a nonlinear distortion component in the desired frequency band, and if further improved by using a linearization mean, the implementation difficulty and complexity of the modulation mode thereof will greatly increase. □ In consideration of the difficulty of hardware implementation, the oversampling frequency of the PWM mode itself is relatively low, and generally in a frequency range of 200 kHz˜400 kHz, which will limit the signal to noise ratio of the coded signal to be further improved due to the oversampling rate.
With the advent of the new generation of ultra wideband digital coded sound source—SACD (Super Audio CD), the PWM coding-based digital power amplifier has been unable to meet the flat frequency response of 2 Hz˜100 kHz required by this digital sound source. In order to ensure the high fidelity reproduction effect of SACD, many experts, scholars and engineers began to develop digital loudspeaker system based on 1-bit Δ-Σ coding, and expected to push the system quantization noise power to a out-band high frequency region by oversampling and noise shaping technology used by Δ-Σ modulation, to improve the tone quality level of the digital system. After many years of research and development accumulation, Japanese company SHARP successfully broke through the technology bottleneck of 1-bit digital amplifier, and since 1998, widely pushed out a series of 1-bit Δ-Σ coding-based digital loudspeaker products one after another in a plurality of acoustic consumption fields.
These 1-bit Δ-Σ coding-based digital loudspeaker systems, only need a simple low-pass filter to complete the digital-to-analog conversion, simple in hardware implementation; the system transfers the noise within an expected audio band to the high frequency region through a high speed switching rate and a 7-order Δ-Σ modulator, to ensure a high fidelity reproduction quality. The 1-bit Δ-Σ coding-based digital loudspeaker system, has the many advantages and meanwhile itself also has the following shortcomings: □ sensitive to the clock jitter, easy to introduce a nonlinear distortion due to the clock jitter; □ in order to maintain the stability of the modulation structure, allowing a very small dynamic range of the input signal; □ requiring a high on-off switching rate, while the power MOSFET transistor will generate many nonlinear distortion components in the process of driving the loudspeaker load to on-off switch in high speed, which also will cause the increased heat, the rised temperature and reduced efficiency of the MOSFET transistor.
In order to solve the defects existing in the 1-bit Δ-Σ coding-based digital loudspeaker system, many scholars turned to research on the multi-bit Δ-Σ coding-based digital systems. The multi-bit Δ-Σ modulation technology overcomes the shortcomings existing in the 1-bit Δ-Σ modulation, meanwhile itself also has a fatal defect—the modulation structure has a high sensitivity to the inconsistency between the frequency responses of the plurality of speaker units (or voice coil units), as well as the separation degree of the spatial locations of the plurality of speaker units, and is easy to introduce a larger coding error due to the inconsistency of the frequency responses of the plurality of unit or the separation of the spatial locations. In addition, the digital power amplifier circuit is easy to be affected by a significant power turbulent wave and fast switching rate effect and to introduce a large nonlinear distortion.
In order to overcome the deviation sensitivity defect of the multi-bit Δ-Σ modulation technology, since 1997, Professor Yasuda Shiaki from Japanese Hosei University and Engineer Okamura Jun from TRIGENCE SEMICONDUCTOR have been cooperating in the development of the multi-bit Δ-Σ coding-based digital system, and proposed a correction method of the system deviation (deviations of the frequency response and the spatial location) based on dynamically mismatch shaping and a beam steering method of a digitalized array based on delay adjustment, and collectively called the Δ-Σ modulation and the dynamic mismatch technologies used by the system as “Dnote” technology; they encapsulated the implementation circuit of “Dnote” technology into an IC chip—“Dnote” chip, and utilized the “Dnote” sample chip to produce a variety of digital loudspeaker system prototype—an 8-unit piezoelectric type linear array loudspeaker system, a 7-unit piezoelectric type ring array system and a 6-voice coil loudspeaker system, which were exhibited in the 2008 Digital Audio Visual Exhibition, and these systems can be driven by a low voltage of 1.5V, without a power amplifier and a LC filter, and has the ability to control the direction. In addition, Mitsui Akihito and Yamada Nobuhito from Janpanese MITSUBISHI COMPANY also filed a patent application (Patent No.: CN 102422650 A) for a Δ-Σ modulation digital loudspeaker on Mar. 10, 2010.
Patent CN102647191 A concerning the “Dnote” technology uses an analog FIR filter and a post-filter, these filters are implemented by employing a switched capacitor filter to achieve; these switched capacitor-based analog filters, are easily affected by external environment, have problems such as voltage drift, temperature drift and noise, and are difficult to achieve a high degree of stability and accuracy.
The drive circuit (or the switching amplifier) mentioned in Patents CN101803401 A, CN 102684700 A, CN 102239706 A, and CN102647191 A concerning “Dnote” technology and Patent CN 102422650 A applied by MITSUBISHI COMPANY, conducts switching operation according to the PDM (Pulse Density Modulation) coding-based digital signal obtained after the treatment of Δ-Σ modulation and mismatch shaping, and moreover, in order to achieve a higher level of signal to noise ratio (SNR), the on-off switching rate of the digital signal processed by Δ-Σ modulating and mismatch shaping is often very high, and is generally required to be more than 10 MHz, for achieving an ideal SNR level, for example, with a 12.5 MHz on-off switching rate, the SNR can reach a level of 100 dB. The over-high on-off switching rate cause a severe decrease of the efficiency of the driving circuit, and meanwhile the over-high switching rate brings an instability into the drive circuit in a slightly higher output power condition, resulting in the driving circuit can not work normally, and for ensuring the stability of the drive circuit in a high speed switching condition, it needs to strictly limit the output power of the drive circuit, and generally for keeping the on-off switching rate of the driving circuit to meet a magnitude order of 10 MHz, the output power of the drive circuit is needed to be limited to a magnitude order of 1 W to ensure the SNR and the harmonic distortion of the driving circuit reach ideal levels, and to keep a steady work of the drive circuit. The drive circuit referred in these patents can not achieve a high power output due to the high speed on-off switching rate limit, and can only be limited to the power output level of 1 W magnitude order.
The mismatch shaping referred in Patents CN101803401 A, CN 102684700 A, CN 102239706 A, and CN102647191 A concerning “Dnote” technology and Patent CN 102422650 A applied by MITSUBISHI COMPANY, does not consider that the input signal amplitude is directly related to the number of the mismatch shaping channel, and does not optimize the channel number participating in mismatch shaping according to the input amplitude of the signal. This defect will cause all channels of the mismatch shaper take part in shaping processing, and consume more energy, and there is optimizing space in reducing the power consumption. The mismatch shaping referred in these patents does not consider the improvement of the SNR brought by increasing the shaping order and optimizing the zero-pole of the shaper.
Patent CN101409560 A mentions that the multiple formats of serial audio signal received from pins of SDATA, BCLK, and LRCK, after serial-to-parallel conversion, are sent to a de-emphasis/interpolation filter, a multi-bit Σ-Δ regulator (DSM), and a dynamic element matching unit (DEM) for combined processing, such that the input data with a high resolution (typically 16 bits to 24 bits) and a low sampling rate (typically 8 KHz to 200 KHz) is transformed into a digital signal with a low resolution (typically 1 bits to 6 bits) and a high sampling frequency (typically 32 times to 128 times of the input frequency); and then the digital signal with a low resolution and a high sampling frequency is transformed by a low pass filter SCF into an analog signal and sent to a sound mixer, by which the analog audio signal output by the digital-to-analog converter is mixed with other analog audio signal, and finally the analog audio signal output by the sound mixer is power amplified, to drive an external headset or speaker to sound. The de-emphasis/interpolation filter, the multi-bit Σ-Δ regulator (DSM), the dynamic element matching unit (DEM) and the low pass filter SCF mentioned in Patent CN101409560 A are typical in the signal processing of a digital-to-analog converter, and this working process does not involve in coding and distribution process directed at multiple units of the loudspeaker array or the multiple voice coil speakers, only processes the input serial audio digital signal digital by digital-to-analog converting to obtain an analog audio output signal, which is sent to a power amplifier and then the amplified signal drives the speaker unit or the headset to sound. The process from power amplifying to outputting of the speaker still belongs to the analog signal transmission process, and the system consisting of the power amplifier and the electro-acoustic transducer of the loudspeaker has a relatively low degree of the integration and relatively low electro-acoustic conversion efficiency, without considering an uniformly digital coding process by taking the multiple speaker units of the speaker array or the multiple voice coils of the multiple voice coil speaker as an entirety.
Aimed at the defects existing in the current digital loudspeaker system devices, and combined with the requirements of low power consumption, small outline, and digitalization and integration development, it is thus desired to find a signal modulation and coding decoding manner with excellent performances and simple implementation, to achieve a digital loudspeaker system device of excellent performances.